U.S. patent number 5,845,251 [Application Number 08/771,792] was granted by the patent office on 1998-12-01 for method, system and product for modifying the bandwidth of subband encoded audio data.
This patent grant is currently assigned to MediaOne Group Inc., U S West, Inc.. Invention is credited to Eliot M. Case.
United States Patent |
5,845,251 |
Case |
December 1, 1998 |
Method, system and product for modifying the bandwidth of subband
encoded audio data
Abstract
A method, system and product are provided for selectively
modifying an encoded audio signal. The method includes receiving
the encoded audio signal, the encoded audio signal having a first
frequency bandwidth, and identifying a delivery point for the
encoded audio signal, the delivery point having a second frequency
bandwidth. The method also includes selecting a plurality of
subbands from the first frequency bandwidth based on the second
frequency bandwidth, and modifying the encoded audio signal based
on the plurality of subbands selected. The system includes control
logic for performing the method. The product includes a storage
medium having computer readable programmed instructions for
performing the method.
Inventors: |
Case; Eliot M. (Denver,
CO) |
Assignee: |
U S West, Inc. (Denver, CO)
MediaOne Group Inc. (Englewood, CO)
|
Family
ID: |
25092988 |
Appl.
No.: |
08/771,792 |
Filed: |
December 20, 1996 |
Current U.S.
Class: |
704/500; 704/501;
G9B/20.001 |
Current CPC
Class: |
G11B
20/00007 (20130101) |
Current International
Class: |
G11B
20/00 (20060101); H04B 001/00 () |
Field of
Search: |
;704/208,278,233,500,501,503,504 ;381/62,119,118 |
References Cited
[Referenced By]
U.S. Patent Documents
Foreign Patent Documents
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0446037A3 |
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Sep 1991 |
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EP |
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0446037A2 |
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Sep 1991 |
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EP |
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0607989A3 |
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Jul 1994 |
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EP |
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0607989A2 |
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Jul 1994 |
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EP |
|
WO91/06945 |
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May 1991 |
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WO |
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WO94/25959 |
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Nov 1994 |
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WO |
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Other References
New Digital Hearing Aids Perk Up Investers' Ears, St. Louis
Post-Dispatch, Sep. 27, 1995. .
Jean-Pierre Renard, Ph.D., B.B.A., High Fidelity Audio Coding, pp.
87-97..
|
Primary Examiner: Hudspeth; David R.
Assistant Examiner: Zintel; Harold
Attorney, Agent or Firm: Brooks & Kushman
Claims
What is claimed is:
1. A method for selectively modifying an encoded audio signal, the
method comprising:
receiving the encoded audio signal, the encoded audio signal having
a first frequency bandwidth;
identifying a delivery point for the encoded audio signal, the
delivery point having a second frequency bandwidth;
selecting a plurality of subbands from the first frequency
bandwidth based on the second frequency bandwidth; and
modifying the encoded audio signal based on the plurality of
subbands selected.
2. The method of claim 1 wherein the second frequency bandwidth is
narrower than the first frequency bandwidth, each of the plurality
of subbands selected has a frequency outside the second frequency
bandwidth, and modifying the encoded audio signal includes
eliminating from the encoded audio signal the plurality of subbands
selected to create a modified encoded audio signal.
3. The method of claim 1 wherein the delivery point is a
transmission network and modifying the encoded audio signal
includes translating a first data format associated with the
encoded audio signal to a second data format associated with the
transmission network.
4. The method of claim 1 wherein the delivery point is a playback
destination and modifying the encoded audio signal includes
translating a first data format associated with the encoded audio
signal to a second data format associated with the playback
destination.
5. The method of claim 1 wherein the delivery point is a storage
medium and modifying the encoded audio signal includes translating
a first data format associated with the encoded audio signal to a
second data format associated with the storage medium.
6. The method of claim 2 wherein the delivery point is a
transmission network, the method further comprising translating a
first data format associated with the modified encoded audio signal
to a second data format associated with the transmission
network.
7. The method of claim 2 wherein the delivery point is a playback
destination, the method further comprising translating a first data
format associated with the modified encoded audio signal to a
second data format associated with the playback destination.
8. The method of claim 2 wherein the delivery point is a storage
medium, the method further comprising translating a first data
format associated with the modified encoded audio signal to a
second data format associated with the storage medium.
9. A system for selectively modifying an encoded audio signal, the
system comprising:
a receiver for receiving the encoded audio signal, the encoded
audio signal having a first frequency bandwidth;
means for identifying a delivery point for the encoded audio
signal, the delivery point having a second frequency bandwidth;
and
control logic operative to select a plurality of subbands from the
first frequency bandwidth based on the second frequency bandwidth,
and modify the encoded audio signal based on the plurality of
subbands selected.
10. The system of claim 9 wherein the second frequency bandwidth is
narrower than the first frequency bandwidth, each of the plurality
of subbands selected has a frequency outside the second frequency
bandwidth, and to modify the encoded audio signal the control logic
is further operative to eliminate from the encoded audio signal the
plurality of subbands selected to create a modified encoded audio
signal.
11. The system of claim 9 wherein the delivery point is a
transmission network and, to modify the encoded audio signal, the
control logic is operative to translate a first data format
associated with the encoded audio signal to a second data format
associated with the transmission network.
12. The system of claim 9 wherein the delivery point is a playback
destination and, to modify the encoded audio signal, the control
logic is operative to translate a first data format associated with
the encoded audio signal to a second data format associated with
the playback destination.
13. The system of claim 9 wherein the delivery point is a storage
medium and, to modify the encoded audio signal, the control logic
is operative to translate a first data format associated with the
encoded audio signal to a second data format associated with the
storage medium.
14. The system of claim 10 wherein the delivery point is a
transmission network and the control logic is further operative to
translate a first data format associated with the modified encoded
audio signal to a second data format associated with the
transmission network.
15. The system of claim 10 wherein the delivery point is a playback
destination and the control logic is further operative to translate
a first data format associated with the modified encoded audio
signal to a second data format associated with the playback
destination.
16. The system of claim 10 wherein the delivery point is a storage
medium and the control logic is further operative to translate a
first data format associated with the modified encoded audio signal
to a second data format associated with the storage medium.
17. A product for selectively modifying an encoded audio signal,
the product comprising a storage medium having computer readable
programmed instructions recorded thereon, the instructions
operative to select a plurality of subbands from a first frequency
bandwidth associated with the encoded audio signal based on a
second frequency bandwidth associated with a delivery point, and
modify the encoded audio signal based on the plurality of subbands
selected.
18. The product of claim 17 wherein the second frequency bandwidth
is narrower than the first frequency bandwidth, each of the
plurality of subbands selected has a frequency outside the second
frequency bandwidth, and to modify the encoded audio signal the
instructions are further operative to eliminate from the encoded
audio signal the plurality of subbands selected to create a
modified encoded audio signal.
19. The product of claim 17 wherein the delivery point is a
transmission network and, to modify the encoded audio signal, the
instructions are operative to translate a first data format
associated with the encoded audio signal to a second data format
associated with the transmission network.
20. The product of claim 17 wherein the delivery point is a
playback destination and, to modify the encoded audio signal, the
instructions are operative to translate a first data format
associated with the encoded audio signal to a second data format
associated with the playback destination.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
This application is related to U.S. patent application Ser. Nos.
08/771,790 entitled "Method, System And Product For Lossless
Encoding Of Digital Audio Data"; 08/771,462 entitled "Method,
System And Product For Modifying The Dynamic Range Of Encoded Audio
Signals"; 08/771,512 entitled "Method, System And Product For
Harmonic Enhancement Of Encoded Audio Signals";08/769,911 entitled
"Method, System And Product For Multiband Compression Of Encoded
Audio Signals";08/777,724 entitled "Method, System And Product For
Mixing Of Encoded Audio Signals"; 08/769,732 entitled "Method,
System And Product For Using Encoded Audio Signals In A Speech
Recognition System"; 08/772,591 entitled "Method, System And
Product For Synthesizing Sound Using Encoded Audio Signals";
08/769,731 entitled "Method, System And Product For Concatenation
Of Sound And Voice Files Using Encoded Audio Data"; and 08/771,469
entitled "Graphic Interface System And Product For Editing Encoded
Audio Data", all of which were filed on the same date and assigned
to the same assignee as the present application.
TECHNICAL FIELD
This invention relates to a method, system and product for
modifying encoded audio data to conform to limited transmission,
storage and/or playback capabilities without creating multiple
source files.
BACKGROUND ART
To more efficiently transmit digital audio data on low bandwidth
data networks, or to store larger amounts of digital audio data in
a small data space, various data compression or encoding systems
and techniques have been developed. Many such encoded audio systems
use as a main element in data reduction the concept of not
transmitting, or otherwise not storing portions of the audio that
might not be perceived by an end user. As a result, such systems
are referred to as perceptually encoded or "lossy" audio
systems.
However, as a result of such data elimination, perceptually encoded
audio systems are not considered "audiophile" quality, and suffer
from processing limitations. To overcome such deficiencies, a
method, system and product have been developed to encode digital
audio signals in a loss-less fashion, which is more properly
referred to as "component audio" rather than perceptual encoding,
since all portions or components of the digital audio signal are
retained. Such a method, system and product are described in detail
in U.S. patent application Ser. No. 08/771,790 entitled "Method,
System And Product For Lossless Encoding Of Digital Audio Data",
which was filed on the same date and assigned to the same assignee
as the present application, and is hereby incorporated by
reference.
To transmit across narrow bandwidth networks or to playback at
narrow bandwidth destinations, however, such encoded audio signals
must first be fully decoded and then converted to the data format
associated with such transmission networks or playback
destinations. Moreover, for encoded audio sound files, multiple
copies thereof must be made off-line and stored for each different
network or destination.
To address these problems, an encoded audio system could be
designed to scale the audio at encoding according to the bandwidth
and data format characteristics of a selected transmission network
or playback destination. However, such a system would not address
dynamically re-scaleable transmission within a given data stream
(i.e., real-time).
Thus, there exists a need for a method, system and product for
modifying transmission and playback of encoded audio signals. Such
a method, system and product would act on a passing data stream to
provide dynamic, real-time conversion between data formats, thereby
eliminating the need for multiple stored copies of a sound asset
for every desired format.
SUMMARY OF THE INVENTION
Accordingly, it is the principle object of the present invention to
provide an improved method, system and product for modifying
transmission and playback of encoded audio signals.
According to the present invention, then, a method is provided for
selectively modifying an encoded audio signal. The method comprises
receiving the encoded audio signal, the encoded audio signal having
a first frequency bandwidth, and identifying a delivery point for
the encoded audio signal, the delivery point having a second
frequency bandwidth. The method further comprises selecting a
plurality of subbands from the first frequency bandwidth based on
the second frequency bandwidth, and modifying the encoded audio
signal based on the plurality of subbands selected.
A system for selectively modifying an encoded audio signal is also
provided. The system comprises a receiver for receiving the encoded
audio signal, the encoded audio signal having a first frequency
bandwidth, and means for identifying a delivery point for the
encoded audio signal, the delivery point having a second frequency
bandwidth. The system further comprises control logic operative to
select a plurality of subbands from the first frequency bandwidth
based on the second frequency bandwidth, and modify the encoded
audio signal based on the plurality of subbands selected.
A product for selectively modifying an encoded audio signal is also
provided. The product comprises a storage medium having computer
readable programmed instructions recorded thereon. The instructions
are operative to select a plurality of subbands from a first
frequency bandwidth associated with the encoded audio signal based
on a second frequency bandwidth associated with a delivery point,
and modify the encoded audio signal based on the plurality of
subbands selected.
These and other objects, features and advantages will be readily
apparent upon consideration of the following detailed description
in conjunction with the accompanying drawings.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is an exemplary encoding format for an audio frame according
to prior art perceptually encoded audio systems;
FIG. 2 is a psychoacoustic model of a human ear including exemplary
masking effects for use with the present invention; and
FIGS. 3a and 3b are simplified block diagrams of the system of the
present invention;
FIGS. 4a, 4b and 4c are graphic representations of original encoded
audio data and exemplary modifications thereto according to the
present invention; and
FIG. 5 is an exemplary storage medium for use with the product of
the present invention.
BEST MODE FOR CARRYING OUT THE INVENTION
Referring now to FIGS. 1-5, the preferred embodiment of the present
invention will now be described. FIG. 1 depicts an exemplary
encoding format for an audio frame according to prior art
perceptually encoded audio systems, such as the various layers of
the Motion Pictures Expert Group (MPEG), Musicam, or others.
Examples of such systems are described in detail in a paper by K.
Brandenburg et al. entitled "ISO-MPEG-1 Audio: A Generic Standard
For Coding High-Quality Digital Audio", Audio Engineering Society,
92nd Convention, Vienna, Austria, March 1992, which is hereby
incorporated by reference.
In that regard, it should be noted that the present invention can
be applied to subband data encoded as either time versus amplitude
(low bit resolution audio bands as in MPEG audio layers 1 or 2, and
Musicam) or as frequency elements representing frequency, phase and
amplitude data (resulting from Fourier transforms or inverse
modified discrete cosine spectral analysis as in MPEG audio layer
3, Dolby AC3 and similar means of spectral analysis) . It should
further be noted that the present invention is suitable for use
with any system using mono, stereo or multichannel sound including
Dolby AC3, 5.1 and 7.1 channel systems.
As seen in FIG. 1, such perceptually encoded digital audio includes
multiple frequency subband data samples (10) , as well as 6 bit
dynamic scale factors (12) (per subband) representing an available
dynamic range of approximately 120 decibels (dB) given a resolution
of 2 dB per scale factor. The bandwidth of each subband is 1/3
octave. Such perceptually encoded digital audio still further
includes a header (14) having information pertaining to sync words
and other system information such as data formats, audio frame
sample rate, channels, etc.
To greatly increase the available dynamic range and/or the
resolution thereof, one or more bits may be added to the dynamic
scale factors (12). For example, by using 8 bit dynamic scale
factors, the dynamic range is doubled to 256 dB and given an
improved 1 dB per scale factor resolution. Alternatively, such 8
bit dynamic scale factors, with a given resolution of 0.5 dB per
scale factor, will provide a dynamic range of 128 dB. In either
case, the accuracy of storage is increased or maintained well
beyond what is needed for dynamic range, while the side-effects of
low resolution dynamic scaling are reduced.
As previously discussed, perceptually encoded audio systems
eliminate portions of the audio that might not be perceived by an
end user. This is accomplished using well known psychoacoustic
modeling of the human ear. Referring now to FIG. 2, such a
psychoacoustic model including exemplary masking effects is shown.
As seen therein, at a given frequency (in kHz), sound levels (in
dB) below the base line curve (40) are inaudible. Using this
information, prior art perceptually encoded audio systems eliminate
data samples in those frequency subbands where the sound level is
likely inaudible.
As also seen therein, short band noise centered at various
frequencies (42, 44, 46, 48) modifies the base line curve (40) to
create what are known as masking effects. That is, such noise (42,
44, 46, 48) raises the level of sound required around such
frequencies before that sound will be audible to the human ear.
Using this information, prior art perceptually encoded audio
systems further eliminate data samples in those frequency subbands
where the sound level is likely inaudible due to such masking
effects.
Alternatively, using a loss-less component audio encoding scheme,
such masked audio may be retained. Once again, such a loss-less
component audio encoding scheme is described in detail in U.S.
patent application Ser. No. 08/771,790 entitled "Method, System And
Product For Lossless Encoding Of Digital Audio Data", which was
filed on the same date and assigned to the same assignee as the
present application, and has been incorporated herein by
reference.
In either case, if no information is present to be encoded into a
subband, the subband does not need to be transmitted. Moreover, if
the subband data is well below the level of audibility (not
including masking effects) , as shown by base line curve (40) of
FIG. 2, the particular subband need not be encoded.
Referring now to FIG. 3a, a simplified block diagram of the system
of the present invention is shown. As seen therein, the system
preferably comprises an appropriately programmed processor (50) for
Digital Signal Processing (DSP) . Processor (50) acts as a receiver
for receiving an encoded audio signal (52) (which may be a stored
sound file/asset) having a frequency bandwidth associated
therewith. In that regard, as previously described, encoded audio
signal (52) may be either a perceptually encoded audio signal or a
component audio signal.
Once programmed, processor (50) provides control logic for
performing various functions of the present invention. In that
regard, processor (50) also receives control input (54) for
identifying any one of a plurality of particular delivery points
(56, 58, 60, 62, 64, 66) for the encoded audio signal (52) Each
delivery point (56, 58, 60, 62, 64, 66) has its own frequency
bandwidth associated therewith. In that regard, delivery points
(56, 58, 60, 62, 64, 66) may be transmission networks, playback
destinations, or storage mediums, and may have any type of data
format including, but not limited to, 8K 8 bit PCM, 6K 4 bit ADPCM,
16 bit 44.1K PCM (including stereo version), component audio, or
perceptually encoded audio such as MPEG (layers 1, 2 or 3),
Musicam, or Real Audio (i.e., internet).
Still referring to FIG. 3a, the control logic of processor (50) is
operative to select a plurality of subbands from the frequency
bandwidth associated with the encoded audio signal (52) based on
the frequency bandwidth associated with the particular delivery
point (56, 58, 60, 62, 64, 66) identified. Where the frequency
bandwidths of the encoded audio signal (52) and the identified
delivery point (56, 58, 60, 62, 64, 66) are the same, the plurality
of frequency subbands selected (68) may be all the frequency
subbands associated with the encoded audio signal (52). This might
be the case where the encoded audio signal (52) has an MPEG layer 2
encoded audio data format and the delivery point has an MPEG layer
3 encoded audio data format, or vice versa. However, where the
frequency bandwidth of the encoded audio signal (52) is greater
than the frequency bandwidth of the delivery point (56, 58, 60, 62,
64, 66) identified, the plurality of frequency subbands selected
are those frequency subbands outside the frequency bandwidth
associated with the identified delivery point (56, 58, 60, 62, 64,
66).
The control logic of processor (50) is further operative to then
modify the encoded audio signal based on the plurality of subbands
selected. To modify the encoded audio signal (52), the control
logic may be operative to directly map/convert/translate/transcode
the data format associated with the encoded audio signal (52) to
the data format associated with the particular delivery point (56,
58, 60, 62, 64, 66)
Alternatively, however, to modify the encoded audio signal (52),
the control logic may be further operative to eliminate from the
encoded audio signal (52) the plurality of subbands selected to
create a modified encoded audio signal (which again may be a sound
file/asset for storage) (70) for continued transmission. In that
case, the control logic of processor (50) is still further
operative to map/convert/translate/transcode the data format
associated with the encoded audio signal (52, 68) or the modified
encoded audio signal (70) to the data format associated with the
delivery point (56, 58, 60, 62, 64, 66) for later decoding and
playback. Once again, such delivery points (56, 58, 60, 62, 64, 66)
may be any transmission networks, playback destinations, or storage
mediums having any data format such as those previously
described.
In such a fashion, the present invention selects portions of the
encoded audio signal (52) for
mapping/conversion/translation/transcoding to the data format of
the delivery point (56, 58, 60, 62, 64, 66). In that regard, the
only portions of the encoded audio signal (52) selected are those
needed based on the bandwidth and data format characteristics of
the delivery point (56, 58, 60, 62, 64, 66).
As is readily apparent, then, processor (50) may also be
represented as shown in the simplified block diagram of FIG. 3b. As
seen therein, and with continuing reference to FIG. 3a, processor
(50) may comprise frequency limiting means (72) for selecting a
plurality of subbands from the frequency bandwidth associated with
the encoded audio signal (52) based on the frequency bandwidth
associated with a particular delivery point (56, 58, 60, 62, 64,
66) , according to the criteria previously described. In this
example, processor (50) also comprises synthesis means (74) for
modifying and/or mapping/converting/translating/transcoding the
encoded audio signal (52), as also previously described.
Referring now to FIGS. 4a-c, graphic representations of original
encoded audio data and exemplary modifications thereto according to
the present invention are shown. In that regard, FIG. 4a depicts
those frequency subbands encoded for an audio signal according to a
32 subband perceptual encoding audio system having a frequency
bandwidth from 20 Hz to 20 kHz. To modify such an encoded audio
signal for transmission and/or playback over an 8K 8 bit PCM
digital phoneline according to the present invention as described
above, only those frequency subbands between 20 Hz and 4 kHz might
be mapped from the encoded audio signal as shown in FIG. 4b.
Similarly, to modify such an encoded audio signal for transmission
and/or playback over another perceptually encoded audio system
having only 16 subbands according to the present invention as
described above, only one-half of the 32 subbands of the original
encoded audio signal might be mapped from the encoded audio signal
as shown in FIG. 4c.
Referring finally to FIG. 5, an exemplary storage medium for the
product of the present invention is shown. In that regard, storage
medium (100) is depicted as a conventional floppy disk, although
any other type of storage medium may also be used.
Storage medium (100) has recorded thereon computer readable
programmed instructions for performing various functions of the
present invention. More particularly, storage medium (100) includes
instructions operative to select a plurality of subbands from a
first frequency bandwidth associated with the encoded audio signal
based on a second frequency bandwidth associated with a delivery
point, and modify the encoded audio signal based on the plurality
of subbands selected.
In that regard, as previously discussed, the second frequency
bandwidth may be narrower than the first frequency bandwidth, and
each of the plurality of subbands selected may have a frequency
outside the second frequency bandwidth. To modify the encoded audio
signal the instructions are then further operative to eliminate
from the encoded audio signal the plurality of subbands selected to
create a modified encoded audio signal.
Further the delivery point may be a transmission network and, to
modify the encoded audio signal, the instructions are operative to
translate a first data format associated with the encoded audio
signal to a second data format associated with the transmission
network. Still further the delivery point may be a playback
destination and, to modify the encoded audio signal, the
instructions are operative to translate a first data format
associated with the encoded audio signal to a second data format
associated with the playback destination.
Thus, by selectively omitting information higher or lower in
frequency than needed for the target application, a given sound
file will be modified and transmitted according to the abilities
required or requested without having multiple copies of the asset
at differing data rates and compression levels. By reading the
header and content specific information within an encoded sound
file (or passing data stream), only the information and data
elements are transmitted that can either be handled by the current
transmission data width or target application, or to decrease
download time, etc. The subtracted information can either be filled
back in at the receiving end to comply with current standard
decoders (such as MPEG layer 1,2,3 etc.) or be constructed into
file formats (or data streams) such as Mu law 8 bit PCM for
telephone lines. The data density can be constantly modulated
during transmission and resynthesized at the receiving end to
whatever conventions are required for use.
In such fashion, faster delivery of sound files across narrow
bandwidth networks is provided, making real-time scaling of data
transmit rates possible. Indeed, transmission speeds are increased
for whatever kind of transport system used, such as internet,
interactive TV, satellite feeds, phonelines, etc. Moreover, only
one source file is necessary for all variable data rates, modes,
formats and qualities of the asset. In that regard, bit rates are
dynamically varied during transmission. Thus, multiple stored
copies of an asset in every desired data format are no longer
necessary.
It should be noted that this invention is designed to act in
real-time on a passing encoded audio data stream at the
distribution level (at the point of transmission or the point of
delivery) and/or as part of a final decoder that reassembles the
signals back to a normal linear audio signal, rather than as part
of the original encoder. In such a fashion, the original program
material can be encoded according to widely deployed encoding
schemes/systems and remain uncompromised. However, the present
invention can also be used for non real-time applications.
In that same regard, it should be noted that the present invention
is suitable for use in any type of DSP application including
computer systems, hearing aids, transmission across networks
including cellular, wireless and cable telephony, internet, cable
television, satellites, audio/video post-production, etc. It should
still further be noted that the present invention can be used in
conjunction with the inventions disclosed in U.S. patent
application Ser. Nos. 08/771,790 entitled "Method, System And
Product For Lossless Encoding Of Digital Audio Data"; 08/771,462
entitled "Method, System And Product For Modifying The Dynamic
Range Of Encoded Audio Signals"; 08/771,512 entitled "Method,
System And Product For Harmonic Enhancement Of Encoded Audio
Signals"; 08/769,911 entitled "Method, System And Product For
Multiband Compression Of Encoded Audio Signals"; 08/777,724
entitled "Method, System And Product For Mixing Of Encoded Audio
Signals"; 08/769,732 entitled "Method, System And Product For Using
Encoded Audio Signals In A Speech Recognition System"; 08/772,591
entitled "Method, System And Product For Synthesizing Sound Using
Encoded Audio Signals"; 08/769,731 entitled "Method, System And
Product For Concatenation Of Sound And Voice Files Using Encoded
Audio Data"; and 08/771,469 entitled "Graphic Interface System And
Product For Editing Encoded Audio Data", all of which were filed on
the same date and assigned to the same assignee as the present
application, and which are hereby incorporated by reference.
As is readily apparent from the foregoing description, then, the
present invention provides a method, system and product for
modifying transmission and playback of encoded audio signals. More
particularly, the present invention acts on a passing data stream
to provide dynamic, real-time conversion between data formats,
thereby eliminating the need for multiple stored copies of a sound
asset for every known format.
It is to be understood that the present invention has been
described above in an illustrative manner and that the terminology
which has been used is intended to be in the nature of words of
description rather than of limitation. As previously stated, many
modifications and variations of the present invention are possible
in light of the above teachings. Therefore, it is also to be
understood that, within the scope of the following claims, the
invention may be practiced otherwise than as specifically described
herein.
* * * * *